Kokoro TTS
RepositoryFreeLightweight 82M parameter open-source TTS with high-quality output.
Capabilities10 decomposed
dual-platform text-to-speech synthesis with 82m parameter neural model
Medium confidenceGenerates natural-sounding speech from text using a lightweight 82-million parameter transformer-based neural model (KModel class) that operates on phoneme sequences rather than raw text, with parallel Python and JavaScript implementations enabling deployment from CLI to web browsers. The KPipeline orchestrates text processing through language-specific G2P conversion (misaki or espeak-ng backends) followed by neural synthesis and ONNX-based audio waveform generation via istftnet modules.
Combines 82M parameter efficiency (vs 1B+ parameter competitors) with dual Python/JavaScript architecture enabling both server and browser deployment; uses misaki + espeak-ng hybrid G2P pipeline for language-agnostic phoneme conversion rather than language-specific models
Smaller model size and Apache 2.0 licensing enable unrestricted commercial deployment where cloud-dependent TTS (Google Cloud, Azure) or GPL-licensed alternatives (Coqui) are impractical; JavaScript support gives browser-native synthesis unavailable in most open-source TTS
language-aware grapheme-to-phoneme conversion with hybrid g2p backends
Medium confidenceConverts text characters to phoneme sequences using a dual-backend architecture: misaki library as primary G2P engine for most languages, with espeak-ng fallback for Hindi and other languages requiring rule-based phonetic conversion. The text processing pipeline (in kokoro/pipeline.py) selects the appropriate G2P backend based on language code, handles text chunking for long inputs, and produces phoneme sequences that feed into neural synthesis.
Hybrid G2P architecture using misaki as primary engine with espeak-ng fallback provides better phonetic accuracy than single-backend approaches; language-specific backend selection (misaki for most, espeak-ng for Hindi) optimizes for each language's phonetic complexity rather than one-size-fits-all approach
More flexible than single-backend G2P (e.g., pure espeak-ng) by combining neural-trained misaki with rule-based espeak-ng; avoids dependency on large language models for phoneme conversion, reducing latency vs LLM-based G2P approaches
onnx-based audio waveform generation from phoneme embeddings
Medium confidenceGenerates raw audio waveforms from phoneme token sequences using ONNX-optimized istftnet modules that perform inverse short-time Fourier transform (ISTFT) synthesis. The KModel class produces mel-spectrogram embeddings from phoneme tokens, which are then converted to linear spectrograms and finally to waveforms via the ONNX-compiled istftnet vocoder, enabling efficient CPU/GPU inference without PyTorch overhead.
Uses ONNX-compiled istftnet vocoder for inference optimization rather than PyTorch-based vocoding, reducing memory footprint and enabling deployment on ONNX Runtime across heterogeneous hardware (CPU, GPU, mobile); istftnet provides direct spectrogram-to-waveform synthesis without intermediate neural vocoder layers
ONNX vocoding is faster than PyTorch-based vocoders (HiFi-GAN, Glow-TTS) on CPU inference; smaller model size than end-to-end neural vocoders enables edge deployment where alternatives require significant computational overhead
multi-voice synthesis with pre-trained voice embeddings
Medium confidenceEnables selection from multiple pre-trained voice styles (e.g., 'af_heart' for American female, various British voices) by conditioning the neural model with voice-specific embeddings. The KModel class accepts a voice identifier parameter that retrieves corresponding embeddings from HuggingFace Hub, which are concatenated with phoneme embeddings during synthesis to produce voice-specific speech characteristics without retraining the base model.
Implements speaker conditioning via pre-trained voice embeddings rather than speaker ID tokens or speaker-specific model variants, enabling voice selection without model duplication; embeddings are downloaded on-demand from HuggingFace Hub rather than bundled, reducing package size
More efficient than maintaining separate model checkpoints per voice (as some TTS systems do); embedding-based conditioning is lighter-weight than speaker encoder networks used in some alternatives, reducing inference latency
python and javascript dual-implementation api with unified semantics
Medium confidenceProvides parallel Python (KPipeline, KModel classes) and JavaScript (KokoroTTS class) implementations with identical functional semantics, enabling code portability and consistent behavior across environments. Both implementations share the same text processing pipeline, model inference logic, and audio synthesis approach, with language-specific optimizations (PyTorch for Python, ONNX.js for JavaScript) while maintaining API compatibility.
Maintains semantic equivalence between Python and JavaScript implementations through shared pipeline design (KPipeline abstraction) rather than transpilation or wrapper layers; both implementations use identical text processing and model inference logic with language-specific runtime optimization
More maintainable than separate Python/JavaScript implementations because core logic is unified; avoids transpilation overhead and complexity of maintaining two codebases with different semantics, unlike some TTS projects with separate Python and JS versions
command-line interface for batch and interactive tts synthesis
Medium confidenceProvides CLI tools for text-to-speech synthesis without programmatic API usage, supporting both interactive input and batch file processing. The CLI wraps the KPipeline class, accepting text input via stdin or file arguments, language/voice parameters, and output file specifications, enabling integration into shell scripts and data processing pipelines.
CLI implementation wraps KPipeline class directly without separate CLI-specific code, maintaining consistency with programmatic API; supports both interactive and batch modes through unified interface
Simpler than cloud-based TTS CLIs (Google Cloud, Azure) because no authentication or API key management required; more accessible than programmatic APIs for non-developers and shell script integration
model export and optimization for production deployment
Medium confidenceProvides utilities (examples/export.py) to export the KModel neural network and istftnet vocoder to ONNX format for optimized inference across different hardware and runtime environments. The export process converts PyTorch models to ONNX intermediate representation, enabling deployment on ONNX Runtime (CPU, GPU, mobile) without PyTorch dependency, reducing model size and inference latency.
Provides explicit export utilities rather than automatic ONNX export, giving developers control over export parameters and optimization settings; separates export from inference, enabling offline optimization workflows
More flexible than automatic export because developers can customize export parameters; avoids runtime overhead of on-demand export compared to systems that export during first inference
streaming audio generation with generator-based processing
Medium confidenceImplements generator-based processing pipeline that yields audio segments incrementally as they are synthesized, rather than buffering entire output. The KPipeline class returns Python generators that yield tuples of (graphemes, phonemes, audio_segment) for each text chunk, enabling memory-efficient processing of long texts and streaming output to audio devices or files.
Uses Python generators to yield audio segments incrementally rather than buffering entire output, enabling memory-efficient processing of arbitrarily long texts; generator pattern provides both phoneme and audio output for each segment, enabling downstream analysis or processing
More memory-efficient than batch processing entire texts; enables real-time streaming output unavailable in systems that require complete synthesis before output; generator pattern is more Pythonic than callback-based streaming
browser-based javascript tts with onnx.js inference
Medium confidenceEnables client-side text-to-speech synthesis in web browsers using ONNX.js runtime for neural model inference and Web Audio API for audio playback. The JavaScript KokoroTTS class implements the same pipeline as Python version but uses ONNX.js for model inference, avoiding server-side processing and enabling offline-capable web applications.
Implements full TTS pipeline in browser using ONNX.js rather than server-side API calls, enabling offline-first web applications; uses Web Audio API for native browser audio playback without external libraries
Eliminates server-side TTS dependency and latency compared to cloud-based TTS APIs; enables offline functionality unavailable in cloud TTS; reduces privacy concerns by keeping audio synthesis local to user's browser
huggingface hub integration for model and voice distribution
Medium confidenceAutomatically downloads and caches model weights, voice embeddings, and language-specific assets from HuggingFace Hub on first use, eliminating manual model management. The system uses HuggingFace's hub_download API to fetch the 82M parameter model, istftnet decoder, voice embeddings, and G2P resources, with local caching to avoid repeated downloads. Model versioning is handled via HuggingFace Hub's revision system, enabling easy updates and rollbacks without code changes.
Integrates HuggingFace Hub for automatic model/voice distribution with transparent caching, eliminating manual model management — most TTS libraries require pre-downloaded model files or manual setup
Simpler than manual model distribution (e.g., downloading from GitHub releases); more flexible than bundling models in packages due to HuggingFace's versioning and update capabilities; reduces deployment friction compared to cloud APIs requiring authentication
Capabilities are decomposed by AI analysis. Each maps to specific user intents and improves with match feedback.
Related Artifactssharing capabilities
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Best For
- ✓Developers building offline-first voice applications with commercial licensing requirements
- ✓Teams deploying TTS to edge devices or resource-constrained servers
- ✓Builders creating multilingual voice interfaces without cloud API costs
- ✓Multilingual TTS applications requiring accurate phonetic representation
- ✓Developers needing phoneme-level control over speech synthesis
- ✓Systems integrating with downstream phoneme-based audio processing
- ✓Production deployments requiring optimized inference performance
- ✓Edge devices and embedded systems with limited computational resources
Known Limitations
- ⚠Supports only 8 languages (American/British English, Spanish, French, Hindi, Italian, Japanese, Brazilian Portuguese, Mandarin Chinese) — no other language support
- ⚠Model inference latency varies by hardware; CPU inference significantly slower than GPU (no benchmarks provided in documentation)
- ⚠No streaming/real-time synthesis — entire text must be processed before audio generation completes
- ⚠Voice selection limited to pre-defined voice embeddings; no voice cloning or custom voice training
- ⚠Phoneme accuracy depends on G2P backend quality; misaki and espeak-ng have different coverage and accuracy profiles
- ⚠No custom phoneme mapping or user-defined phoneme inventories
Requirements
Input / Output
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About
Lightweight open-source text-to-speech model delivering high-quality speech synthesis with only 82 million parameters, supporting American and British English with multiple voice styles and Apache 2.0 licensing for unrestricted commercial use.
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